hello List,
i am in a situation where i cannot get cdr records for call originated from
CLI , i am not able to get when i used application or extension.
is there any solution regarding this ,i working since last 3 days onto this.
regards
Dhaval
--
Hi Dhaval,
I 'm in the almost same situation.
I've already post a issue with asterisk.
https://issues.asterisk.org/view.php?id=17826
Is you only use an originate and not an originate en then redial maybe this
link helps you further.
https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes
Still I am also facing the call disconnection when there is a third call. I
am using Netmod BRI router and the output of the BRI router lines are
connected to FXO ports in Asterisk.
Where in Asterisk I am facing the call disconnection when there is a third
call..
On Tue, Sep 28, 2010 at 4:22 PM,
Hi All;
Did anyone try to implement (installation and configuration and running) for
more than one asterisk instance (two or three instances), where each asterisk
instance to work on a difference IP than the other where the server already has
more than one IP address.
We need to implement
On Tue, 5 Oct 2010, bilal ghayyad wrote:
Hi All;
Did anyone try to implement (installation and configuration and running)
for more than one asterisk instance (two or three instances), where each
asterisk instance to work on a difference IP than the other where the
server already has more
Hi Arjan,
i am able to solve this problem after adding this patch and adding
unanswered=yes onto cdr.conf
https://issues.asterisk.org/file_download.php?file_id=24431type=bug
regards
Dhaval
On Tue, Oct 5, 2010 at 1:12 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Hi Dhaval,
Hi
We can run multiple instance of asterisk in same box with different IP
and port. U need to install asterisk in different location eg: 1:
/home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path ,
listen ip and port should be different.
command to run:
$
On Tue, 5 Oct 2010, Nikhil wrote:
Hi
We can run multiple instance of asterisk in same box with different IP
and port. U need to install asterisk in different location eg: 1:
/home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path ,
listen ip and port should be different.
On 5 Oct 2010, at 15:13, Gordon Henderson wrote:
$ /home/asterisk1/usr/sbin/asterisk -g for first asterisk
$ /home/asterisk2/usr/sbin/asterisk -g for second asterisk
I did that before I moved to LXC, but you can't use the standard port 5060
for all instances, only one - might be OK in
Any pointers on this one?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 04, 2010 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Registering
On Tue, 5 Oct 2010, Steve Howes wrote:
On 5 Oct 2010, at 15:13, Gordon Henderson wrote:
$ /home/asterisk1/usr/sbin/asterisk -g for first asterisk
$ /home/asterisk2/usr/sbin/asterisk -g for second asterisk
I did that before I moved to LXC, but you can't use the standard port 5060
for all
You can use proxmox from proxmox.com. I am using it for the same reason you
want to use it. I have been testing it for some time now and it works great.
Proxmox is an excellent hypervisor and it is free. Easy to install and
simple to setup. Install it drom its ISO. Then you can download a OenVZ
On 03/10/10 21:19, Greg Saunders wrote:
Hello all. I was recently the victim of a SIP flood attack. I'm
wondering what is the best method to prevent such things in the future.
Many thanks
Greg
do one of the following:
- use deny permit lines in sip.conf /or iax.conf to restrict any
remote
On 02/10/10 17:24, mancyb...@gmail.com wrote:
Hi All,
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock
timing source for conferences to work properly ?
Maybe the Digium TDM410PLF card
without any daughter card
would do
http://www.ip-phone-forum.de/showthread.php?t=188877
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Tue,
Hello my friend Ingmar,
I would like to know the cable you used? how was the connection? i'm using
this one:
http://wiki.sangoma.com/Pinouts#A108 Loop Back
Is this ok? what should i do my friend, my problems are understand the
fisicall connection :(
Best Regards!!!
2010/9/24 Ingmar Steen
Has anyone a solution for me
- with Meetme(,Ms)asterisk plays conf-invalid if a room not exist
- with Meetme(123,Ms) asterisk plays not conf-invalid if the room not exist
and asterisk hangup
I am happy about any proposal.
Thanks
Daniel
--
On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.com wrote:
Hello my friend Ingmar,
I would like to know the cable you used? how was the connection? i'm using
this one:
http://wiki.sangoma.com/Pinouts#A108 Loop Back
Is this ok? what should i do my friend, my problems are
Hello,
I'm trying to configure Asterisk with Radius cdr support.
Asterisk version 1.6.2.13
Server Radius: Freeradius version 1.X
Radius client: radiusclient-ng version 0.5.5
With the Asterisk core debug on 1 when a call terminate, on the console
appear this error:
Unable to create RADIUS
On Tue, 05 Oct 2010 17:30:49 +0100
Paul Hayes p...@provu.co.uk wrote:
On 02/10/10 17:24, mancyb...@gmail.com wrote:
Hi All,
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock
timing source for conferences to work
I now have an OpenVox A400P and it is working well. Thanks to Ade
Vickers for the recommendation, which I second.
However, I need to make a slow transition between a conventional
multiple-extension setup and a full VoIP network on these premises. So
at the moment the Asterisk box shares the PSTN
Hi list,
I was wondering if anyone had any solution to either one of two issues
I'm having:
I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware,
it works very well for the most part, but after less then a week of
heavy usage, eventually the phone gets into a state where it
Hi Mike,
Which is the real name for this peer?
If you want look the configuration peer on Asterisk console try:
CLI sip show peer accountname load
To register to this account on Ekiga... accountname is the name of the
extensions you have to configure.
BR
- Andrea
--
Hi Mike,
Which is the real name for this peer?
If you want look the configuration peer on Asterisk console try:
CLI sip show peer accountname load
To register to this account on Ekiga... accountname is the name of the
extensions you have to configure.
BR
- Andrea
--
On 5 October 2010 21:16, bakko asannu...@gmail.com wrote:
Hello,
I'm trying to configure Asterisk with Radius cdr support.
Asterisk version 1.6.2.13
Server Radius: Freeradius version 1.X
Radius client: radiusclient-ng version 0.5.5
With the Asterisk core debug on 1 when a call terminate,
Hi,
Have you got a dictionary file with the attributes for asterisk?
Yes, my radiusclient-ng dictionary include dictionary.digium
BR
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
I know this doesn't answer your question directly, but Where are you getting
the Sip 9.0 software? It is not available on Cisco's website.
I have Sip 8.9 on my phone and I have noticed that after about 45 mins on a
call it will hang up and drop the desktop connection that runs through the
phone.
Hi there!
I am trying to configure Web-meetme on Asterisk 1.6. I have followed the
README and everything looks ok,therefore, when I try to open the webpage appear
the folowing messages:DB Error: connect failed Testing with a php script, the
message Connected successfully is shown. Att,
On 10/4/10 12:27 PM, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input,
On 10/3/10 11:20 AM, Daniel Knoll wrote:
Hello,
is it possible to check more than one condition for GOTOIF in the dialplan?
yes. check out asterisk expressions on wiki pages
http://www.voip-info.org/wiki/view/Asterisk+Expressions
--
Edwin Lam edwin@officegeneral.com
Systems Engineer,
Carlos Chavez wrote:
On Mon, 2010-10-04 at 14:27 -0500, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
Hi,
As per the Asterisk documentation mentioned in the
http://svnview.digium.com/svn/asterisk/ ... iew=markup followed the
procedure for Call Detail Recording to RADIUS Server.
I was getting the following error DEBUG[12542] cdr_radius.c: Unable to create
RADIUS record. CDR not recorded!
Any
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